[Asterisk-code-review] rtp_range: (testsuite[16])
Michael Bradeen
asteriskteam at digium.com
Tue Oct 26 15:36:45 CDT 2021
Michael Bradeen has uploaded this change for review. ( https://gerrit.asterisk.org/c/testsuite/+/16636 )
Change subject: rtp_range:
......................................................................
rtp_range:
New set of tests to verify that the rtp range configured in rtp.conf
is not exceeded by the port search logic and that asterisk does not
get into an infinite loop while searching for a free port.
Four tests are added based on the start and stop port parity.
rtp_range_even_even: begining and end of range are even
rtp_range_even_odd: start port is even, end port is odd
rtp_range_odd_even: start port is odd, end port is even
rtp_range_odd_odd: start port is odd, end port is odd
In each test we expect three calls to pass then three more to fail,
rejected with a 488 by asterisk due to it being out of free ports.
ASTERISK-27406
Change-Id: Iefa8545cc1785b1e32a42d740f8f567706c73199
---
A tests/rtp/rtp_range/rtp_range_even_even/configs/ast1/extensions.conf
A tests/rtp/rtp_range/rtp_range_even_even/configs/ast1/pjsip.conf
A tests/rtp/rtp_range/rtp_range_even_even/configs/ast1/rtp.conf
A tests/rtp/rtp_range/rtp_range_even_even/sipp/call_accept_test_range.xml
A tests/rtp/rtp_range/rtp_range_even_even/test-config.yaml
A tests/rtp/rtp_range/rtp_range_even_odd/configs/ast1/extensions.conf
A tests/rtp/rtp_range/rtp_range_even_odd/configs/ast1/pjsip.conf
A tests/rtp/rtp_range/rtp_range_even_odd/configs/ast1/rtp.conf
A tests/rtp/rtp_range/rtp_range_even_odd/sipp/call_accept_test_range.xml
A tests/rtp/rtp_range/rtp_range_even_odd/test-config.yaml
A tests/rtp/rtp_range/rtp_range_odd_even/configs/ast1/extensions.conf
A tests/rtp/rtp_range/rtp_range_odd_even/configs/ast1/pjsip.conf
A tests/rtp/rtp_range/rtp_range_odd_even/configs/ast1/rtp.conf
A tests/rtp/rtp_range/rtp_range_odd_even/sipp/call_accept_test_range.xml
A tests/rtp/rtp_range/rtp_range_odd_even/test-config.yaml
A tests/rtp/rtp_range/rtp_range_odd_odd/configs/ast1/extensions.conf
A tests/rtp/rtp_range/rtp_range_odd_odd/configs/ast1/pjsip.conf
A tests/rtp/rtp_range/rtp_range_odd_odd/configs/ast1/rtp.conf
A tests/rtp/rtp_range/rtp_range_odd_odd/sipp/call_accept_test_range.xml
A tests/rtp/rtp_range/rtp_range_odd_odd/test-config.yaml
A tests/rtp/rtp_range/tests.yaml
M tests/rtp/tests.yaml
22 files changed, 703 insertions(+), 0 deletions(-)
git pull ssh://gerrit.asterisk.org:29418/testsuite refs/changes/36/16636/1
diff --git a/tests/rtp/rtp_range/rtp_range_even_even/configs/ast1/extensions.conf b/tests/rtp/rtp_range/rtp_range_even_even/configs/ast1/extensions.conf
new file mode 100644
index 0000000..87291fb
--- /dev/null
+++ b/tests/rtp/rtp_range/rtp_range_even_even/configs/ast1/extensions.conf
@@ -0,0 +1,6 @@
+[default]
+
+exten => test,1,Answer
+exten => test,n,Noop(RTCP Range Test ASTERISK-27406)
+exten => test,n,Wait(15)
+exten => test,n,Hangup
diff --git a/tests/rtp/rtp_range/rtp_range_even_even/configs/ast1/pjsip.conf b/tests/rtp/rtp_range/rtp_range_even_even/configs/ast1/pjsip.conf
new file mode 100644
index 0000000..bef6c2f
--- /dev/null
+++ b/tests/rtp/rtp_range/rtp_range_even_even/configs/ast1/pjsip.conf
@@ -0,0 +1,19 @@
+[global]
+type=global
+debug=yes
+
+[local]
+type=transport
+bind=0.0.0.0
+protocol=udp
+
+[alice]
+type=endpoint
+context=default
+allow=!all,ulaw
+transport=local
+direct_media=no
+
+[alice]
+type=aor
+contact=sip:alice at 127.0.0.1:5061
diff --git a/tests/rtp/rtp_range/rtp_range_even_even/configs/ast1/rtp.conf b/tests/rtp/rtp_range/rtp_range_even_even/configs/ast1/rtp.conf
new file mode 100644
index 0000000..5472994
--- /dev/null
+++ b/tests/rtp/rtp_range/rtp_range_even_even/configs/ast1/rtp.conf
@@ -0,0 +1,3 @@
+[general]
+rtpstart=10000
+rtpend=10004
diff --git a/tests/rtp/rtp_range/rtp_range_even_even/sipp/call_accept_test_range.xml b/tests/rtp/rtp_range/rtp_range_even_even/sipp/call_accept_test_range.xml
new file mode 100644
index 0000000..68276a5
--- /dev/null
+++ b/tests/rtp/rtp_range/rtp_range_even_even/sipp/call_accept_test_range.xml
@@ -0,0 +1,117 @@
+<?xml version="1.0" encoding="ISO-8859-1" ?>
+<!DOCTYPE scenario SYSTEM "sipp.dtd">
+
+<scenario name="INVITE to echo with SDP in initial INVITE">
+ <send retrans="500">
+ <![CDATA[
+
+ INVITE sip:test@[remote_ip]:[remote_port] SIP/2.0
+ Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
+ From: test[call_number] <sip:[service]@[local_ip]:[local_port]>;tag=[call_number]
+ To: test <sip:test@[remote_ip]:[remote_port]>
+ Call-ID: [call_id]
+ CSeq: 1 INVITE
+ Contact: <sip:test@[local_ip]:[local_port];transport=[transport]>
+ Max-Forwards: 70
+ Subject: Test
+ User-Agent: Test
+ Content-Type: application/sdp
+ Content-Length: [len]
+
+ v=0
+ o=phoneA 53655765 2353687637 IN IP[local_ip_type] [local_ip]
+ s=-
+ c=IN IP[media_ip_type] [media_ip]
+ t=0 0
+ m=audio 6000 RTP/AVP 0
+ a=rtpmap:0 PCMU/8000
+
+ ]]>
+ </send>
+
+ <recv response="100"
+ optional="true">
+ </recv>
+
+ <recv response="488"
+ optional="true"
+ next="1"
+ rtd="true">
+ <action>
+ <ereg regexp="test[4-6]" search_in="hdr" header="From: " assign_to="4"/>
+ <test assign_to="4" variable="4" compare="equal" value=""/>
+ </action>
+ </recv>
+
+ <recv response="200" rtd="true">
+ <action>
+ <ereg regexp="m=audio 1000[0-4] RTP/AVP 0+..*"
+ search_in="body" check_it="true" assign_to="5"/>
+ <test assign_to="5" variable="5" compare="equal" value=""/>
+ </action>
+ </recv>
+
+ <send>
+ <![CDATA[
+
+ ACK sip:test@[remote_ip]:[remote_port] SIP/2.0
+ Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
+ From: test[call_number] <sip:[service]@[local_ip]:[local_port]>;tag=[call_number]
+ To: test <sip:test@[remote_ip]:[remote_port]>[peer_tag_param]
+ Call-ID: [call_id]
+ CSeq: 1 ACK
+ Contact: <sip:test@[local_ip]:[local_port];transport=[transport]>
+ Max-Forwards: 70
+ Subject: Test
+ Content-Length: 0
+
+ ]]>
+ </send>
+
+ <recv request="BYE">
+ </recv>
+
+ <send next="2">
+ <![CDATA[
+
+ SIP/2.0 200 OK
+ [last_Via:]
+ [last_From:]
+ [last_To:]
+ [last_Call-ID:]
+ [last_CSeq:]
+ Contact: <sip:[local_ip]:[local_port];transport=[transport]>
+ Content-Length: 0
+
+ ]]>
+ </send>
+
+ <label id="1"/>
+
+ <send next="2">
+ <![CDATA[
+
+ ACK sip:test@[remote_ip]:[remote_port] SIP/2.0
+ Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
+ From: test[call_number] <sip:[service]@[local_ip]:[local_port]>;tag=[call_number]
+ To: test <sip:test@[remote_ip]:[remote_port]>[peer_tag_param]
+ Call-ID: [call_id]
+ CSeq: 1 ACK
+ Contact: <sip:test@[local_ip]:[local_port];transport=[transport]>
+ Max-Forwards: 70
+ Subject: Test
+ Content-Length: 0
+
+ ]]>
+ </send>
+
+ <label id="2"/>
+
+ <!-- definition of the response time repartition table (unit is ms) -->
+ <ResponseTimeRepartition value="10, 20, 30, 40, 50, 100, 150, 200"/>
+
+ <!-- definition of the call length repartition table (unit is ms) -->
+ <CallLengthRepartition value="10, 50, 100, 500, 1000, 5000, 10000"/>
+
+</scenario>
+
diff --git a/tests/rtp/rtp_range/rtp_range_even_even/test-config.yaml b/tests/rtp/rtp_range/rtp_range_even_even/test-config.yaml
new file mode 100644
index 0000000..8256506
--- /dev/null
+++ b/tests/rtp/rtp_range/rtp_range_even_even/test-config.yaml
@@ -0,0 +1,29 @@
+testinfo:
+ summary: 'Verify Asterisk does not go inton an infinite loop and uses correct range'
+ description: |
+ This test verifies that the correct RTP Range is used and that Asterisk does not go
+ in to an infinite loop searching for a free RTP port
+ issues:
+ - jira: 'ASTERISK-27406'
+
+test-modules:
+ add-test-to-search-path: True
+ test-object:
+ config-section: 'test-object-config'
+ typename: 'sipp.SIPpTestCase'
+
+
+test-object-config:
+ memcheck-delay-stop: 7
+ test-iterations:
+ - scenarios:
+ - { 'key-args': {'scenario': 'call_accept_test_range.xml', '-i': '127.0.0.1', '-p': '5061', '-s': 'alice', '-timeout': '20s', '-r': '1', '-m': '6', '-l': '6'} }
+properties:
+ dependencies:
+ - sipp :
+ version : 'v3.0'
+ - asterisk : 'chan_pjsip'
+
+ tags:
+ - pjsip
+ - RTP
\ No newline at end of file
diff --git a/tests/rtp/rtp_range/rtp_range_even_odd/configs/ast1/extensions.conf b/tests/rtp/rtp_range/rtp_range_even_odd/configs/ast1/extensions.conf
new file mode 100644
index 0000000..87291fb
--- /dev/null
+++ b/tests/rtp/rtp_range/rtp_range_even_odd/configs/ast1/extensions.conf
@@ -0,0 +1,6 @@
+[default]
+
+exten => test,1,Answer
+exten => test,n,Noop(RTCP Range Test ASTERISK-27406)
+exten => test,n,Wait(15)
+exten => test,n,Hangup
diff --git a/tests/rtp/rtp_range/rtp_range_even_odd/configs/ast1/pjsip.conf b/tests/rtp/rtp_range/rtp_range_even_odd/configs/ast1/pjsip.conf
new file mode 100644
index 0000000..bef6c2f
--- /dev/null
+++ b/tests/rtp/rtp_range/rtp_range_even_odd/configs/ast1/pjsip.conf
@@ -0,0 +1,19 @@
+[global]
+type=global
+debug=yes
+
+[local]
+type=transport
+bind=0.0.0.0
+protocol=udp
+
+[alice]
+type=endpoint
+context=default
+allow=!all,ulaw
+transport=local
+direct_media=no
+
+[alice]
+type=aor
+contact=sip:alice at 127.0.0.1:5061
diff --git a/tests/rtp/rtp_range/rtp_range_even_odd/configs/ast1/rtp.conf b/tests/rtp/rtp_range/rtp_range_even_odd/configs/ast1/rtp.conf
new file mode 100644
index 0000000..9c05a59
--- /dev/null
+++ b/tests/rtp/rtp_range/rtp_range_even_odd/configs/ast1/rtp.conf
@@ -0,0 +1,3 @@
+[general]
+rtpstart=10000
+rtpend=10005
diff --git a/tests/rtp/rtp_range/rtp_range_even_odd/sipp/call_accept_test_range.xml b/tests/rtp/rtp_range/rtp_range_even_odd/sipp/call_accept_test_range.xml
new file mode 100644
index 0000000..68276a5
--- /dev/null
+++ b/tests/rtp/rtp_range/rtp_range_even_odd/sipp/call_accept_test_range.xml
@@ -0,0 +1,117 @@
+<?xml version="1.0" encoding="ISO-8859-1" ?>
+<!DOCTYPE scenario SYSTEM "sipp.dtd">
+
+<scenario name="INVITE to echo with SDP in initial INVITE">
+ <send retrans="500">
+ <![CDATA[
+
+ INVITE sip:test@[remote_ip]:[remote_port] SIP/2.0
+ Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
+ From: test[call_number] <sip:[service]@[local_ip]:[local_port]>;tag=[call_number]
+ To: test <sip:test@[remote_ip]:[remote_port]>
+ Call-ID: [call_id]
+ CSeq: 1 INVITE
+ Contact: <sip:test@[local_ip]:[local_port];transport=[transport]>
+ Max-Forwards: 70
+ Subject: Test
+ User-Agent: Test
+ Content-Type: application/sdp
+ Content-Length: [len]
+
+ v=0
+ o=phoneA 53655765 2353687637 IN IP[local_ip_type] [local_ip]
+ s=-
+ c=IN IP[media_ip_type] [media_ip]
+ t=0 0
+ m=audio 6000 RTP/AVP 0
+ a=rtpmap:0 PCMU/8000
+
+ ]]>
+ </send>
+
+ <recv response="100"
+ optional="true">
+ </recv>
+
+ <recv response="488"
+ optional="true"
+ next="1"
+ rtd="true">
+ <action>
+ <ereg regexp="test[4-6]" search_in="hdr" header="From: " assign_to="4"/>
+ <test assign_to="4" variable="4" compare="equal" value=""/>
+ </action>
+ </recv>
+
+ <recv response="200" rtd="true">
+ <action>
+ <ereg regexp="m=audio 1000[0-4] RTP/AVP 0+..*"
+ search_in="body" check_it="true" assign_to="5"/>
+ <test assign_to="5" variable="5" compare="equal" value=""/>
+ </action>
+ </recv>
+
+ <send>
+ <![CDATA[
+
+ ACK sip:test@[remote_ip]:[remote_port] SIP/2.0
+ Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
+ From: test[call_number] <sip:[service]@[local_ip]:[local_port]>;tag=[call_number]
+ To: test <sip:test@[remote_ip]:[remote_port]>[peer_tag_param]
+ Call-ID: [call_id]
+ CSeq: 1 ACK
+ Contact: <sip:test@[local_ip]:[local_port];transport=[transport]>
+ Max-Forwards: 70
+ Subject: Test
+ Content-Length: 0
+
+ ]]>
+ </send>
+
+ <recv request="BYE">
+ </recv>
+
+ <send next="2">
+ <![CDATA[
+
+ SIP/2.0 200 OK
+ [last_Via:]
+ [last_From:]
+ [last_To:]
+ [last_Call-ID:]
+ [last_CSeq:]
+ Contact: <sip:[local_ip]:[local_port];transport=[transport]>
+ Content-Length: 0
+
+ ]]>
+ </send>
+
+ <label id="1"/>
+
+ <send next="2">
+ <![CDATA[
+
+ ACK sip:test@[remote_ip]:[remote_port] SIP/2.0
+ Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
+ From: test[call_number] <sip:[service]@[local_ip]:[local_port]>;tag=[call_number]
+ To: test <sip:test@[remote_ip]:[remote_port]>[peer_tag_param]
+ Call-ID: [call_id]
+ CSeq: 1 ACK
+ Contact: <sip:test@[local_ip]:[local_port];transport=[transport]>
+ Max-Forwards: 70
+ Subject: Test
+ Content-Length: 0
+
+ ]]>
+ </send>
+
+ <label id="2"/>
+
+ <!-- definition of the response time repartition table (unit is ms) -->
+ <ResponseTimeRepartition value="10, 20, 30, 40, 50, 100, 150, 200"/>
+
+ <!-- definition of the call length repartition table (unit is ms) -->
+ <CallLengthRepartition value="10, 50, 100, 500, 1000, 5000, 10000"/>
+
+</scenario>
+
diff --git a/tests/rtp/rtp_range/rtp_range_even_odd/test-config.yaml b/tests/rtp/rtp_range/rtp_range_even_odd/test-config.yaml
new file mode 100644
index 0000000..a1d29d9
--- /dev/null
+++ b/tests/rtp/rtp_range/rtp_range_even_odd/test-config.yaml
@@ -0,0 +1,29 @@
+testinfo:
+ summary: 'Verify Asterisk does not go inton an infinite loop and uses correct range'
+ description: |
+ This test verifies that the correct RTP Range is used and that Asterisk does not go
+ in to an infinite loop searching for a free RTP port
+ issues:
+ - jira: 'ASTERISK-27406'
+
+test-modules:
+ add-test-to-search-path: True
+ test-object:
+ config-section: 'test-object-config'
+ typename: 'sipp.SIPpTestCase'
+
+
+test-object-config:
+ memcheck-delay-stop: 7
+ test-iterations:
+ - scenarios:
+ - { 'key-args': {'scenario': 'call_accept_test_range.xml', '-i': '127.0.0.1', '-p': '5061', '-s': 'alice', '-timeout': '20s', '-r': '1', '-m': '6', '-l': '6'} }
+properties:
+ dependencies:
+ - sipp :
+ version : 'v3.0'
+ - asterisk : 'chan_pjsip'
+
+ tags:
+ - pjsip
+ - RTP
\ No newline at end of file
diff --git a/tests/rtp/rtp_range/rtp_range_odd_even/configs/ast1/extensions.conf b/tests/rtp/rtp_range/rtp_range_odd_even/configs/ast1/extensions.conf
new file mode 100644
index 0000000..87291fb
--- /dev/null
+++ b/tests/rtp/rtp_range/rtp_range_odd_even/configs/ast1/extensions.conf
@@ -0,0 +1,6 @@
+[default]
+
+exten => test,1,Answer
+exten => test,n,Noop(RTCP Range Test ASTERISK-27406)
+exten => test,n,Wait(15)
+exten => test,n,Hangup
diff --git a/tests/rtp/rtp_range/rtp_range_odd_even/configs/ast1/pjsip.conf b/tests/rtp/rtp_range/rtp_range_odd_even/configs/ast1/pjsip.conf
new file mode 100644
index 0000000..bef6c2f
--- /dev/null
+++ b/tests/rtp/rtp_range/rtp_range_odd_even/configs/ast1/pjsip.conf
@@ -0,0 +1,19 @@
+[global]
+type=global
+debug=yes
+
+[local]
+type=transport
+bind=0.0.0.0
+protocol=udp
+
+[alice]
+type=endpoint
+context=default
+allow=!all,ulaw
+transport=local
+direct_media=no
+
+[alice]
+type=aor
+contact=sip:alice at 127.0.0.1:5061
diff --git a/tests/rtp/rtp_range/rtp_range_odd_even/configs/ast1/rtp.conf b/tests/rtp/rtp_range/rtp_range_odd_even/configs/ast1/rtp.conf
new file mode 100644
index 0000000..55f467a
--- /dev/null
+++ b/tests/rtp/rtp_range/rtp_range_odd_even/configs/ast1/rtp.conf
@@ -0,0 +1,3 @@
+[general]
+rtpstart=9999
+rtpend=10004
diff --git a/tests/rtp/rtp_range/rtp_range_odd_even/sipp/call_accept_test_range.xml b/tests/rtp/rtp_range/rtp_range_odd_even/sipp/call_accept_test_range.xml
new file mode 100644
index 0000000..68276a5
--- /dev/null
+++ b/tests/rtp/rtp_range/rtp_range_odd_even/sipp/call_accept_test_range.xml
@@ -0,0 +1,117 @@
+<?xml version="1.0" encoding="ISO-8859-1" ?>
+<!DOCTYPE scenario SYSTEM "sipp.dtd">
+
+<scenario name="INVITE to echo with SDP in initial INVITE">
+ <send retrans="500">
+ <![CDATA[
+
+ INVITE sip:test@[remote_ip]:[remote_port] SIP/2.0
+ Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
+ From: test[call_number] <sip:[service]@[local_ip]:[local_port]>;tag=[call_number]
+ To: test <sip:test@[remote_ip]:[remote_port]>
+ Call-ID: [call_id]
+ CSeq: 1 INVITE
+ Contact: <sip:test@[local_ip]:[local_port];transport=[transport]>
+ Max-Forwards: 70
+ Subject: Test
+ User-Agent: Test
+ Content-Type: application/sdp
+ Content-Length: [len]
+
+ v=0
+ o=phoneA 53655765 2353687637 IN IP[local_ip_type] [local_ip]
+ s=-
+ c=IN IP[media_ip_type] [media_ip]
+ t=0 0
+ m=audio 6000 RTP/AVP 0
+ a=rtpmap:0 PCMU/8000
+
+ ]]>
+ </send>
+
+ <recv response="100"
+ optional="true">
+ </recv>
+
+ <recv response="488"
+ optional="true"
+ next="1"
+ rtd="true">
+ <action>
+ <ereg regexp="test[4-6]" search_in="hdr" header="From: " assign_to="4"/>
+ <test assign_to="4" variable="4" compare="equal" value=""/>
+ </action>
+ </recv>
+
+ <recv response="200" rtd="true">
+ <action>
+ <ereg regexp="m=audio 1000[0-4] RTP/AVP 0+..*"
+ search_in="body" check_it="true" assign_to="5"/>
+ <test assign_to="5" variable="5" compare="equal" value=""/>
+ </action>
+ </recv>
+
+ <send>
+ <![CDATA[
+
+ ACK sip:test@[remote_ip]:[remote_port] SIP/2.0
+ Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
+ From: test[call_number] <sip:[service]@[local_ip]:[local_port]>;tag=[call_number]
+ To: test <sip:test@[remote_ip]:[remote_port]>[peer_tag_param]
+ Call-ID: [call_id]
+ CSeq: 1 ACK
+ Contact: <sip:test@[local_ip]:[local_port];transport=[transport]>
+ Max-Forwards: 70
+ Subject: Test
+ Content-Length: 0
+
+ ]]>
+ </send>
+
+ <recv request="BYE">
+ </recv>
+
+ <send next="2">
+ <![CDATA[
+
+ SIP/2.0 200 OK
+ [last_Via:]
+ [last_From:]
+ [last_To:]
+ [last_Call-ID:]
+ [last_CSeq:]
+ Contact: <sip:[local_ip]:[local_port];transport=[transport]>
+ Content-Length: 0
+
+ ]]>
+ </send>
+
+ <label id="1"/>
+
+ <send next="2">
+ <![CDATA[
+
+ ACK sip:test@[remote_ip]:[remote_port] SIP/2.0
+ Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
+ From: test[call_number] <sip:[service]@[local_ip]:[local_port]>;tag=[call_number]
+ To: test <sip:test@[remote_ip]:[remote_port]>[peer_tag_param]
+ Call-ID: [call_id]
+ CSeq: 1 ACK
+ Contact: <sip:test@[local_ip]:[local_port];transport=[transport]>
+ Max-Forwards: 70
+ Subject: Test
+ Content-Length: 0
+
+ ]]>
+ </send>
+
+ <label id="2"/>
+
+ <!-- definition of the response time repartition table (unit is ms) -->
+ <ResponseTimeRepartition value="10, 20, 30, 40, 50, 100, 150, 200"/>
+
+ <!-- definition of the call length repartition table (unit is ms) -->
+ <CallLengthRepartition value="10, 50, 100, 500, 1000, 5000, 10000"/>
+
+</scenario>
+
diff --git a/tests/rtp/rtp_range/rtp_range_odd_even/test-config.yaml b/tests/rtp/rtp_range/rtp_range_odd_even/test-config.yaml
new file mode 100644
index 0000000..8256506
--- /dev/null
+++ b/tests/rtp/rtp_range/rtp_range_odd_even/test-config.yaml
@@ -0,0 +1,29 @@
+testinfo:
+ summary: 'Verify Asterisk does not go inton an infinite loop and uses correct range'
+ description: |
+ This test verifies that the correct RTP Range is used and that Asterisk does not go
+ in to an infinite loop searching for a free RTP port
+ issues:
+ - jira: 'ASTERISK-27406'
+
+test-modules:
+ add-test-to-search-path: True
+ test-object:
+ config-section: 'test-object-config'
+ typename: 'sipp.SIPpTestCase'
+
+
+test-object-config:
+ memcheck-delay-stop: 7
+ test-iterations:
+ - scenarios:
+ - { 'key-args': {'scenario': 'call_accept_test_range.xml', '-i': '127.0.0.1', '-p': '5061', '-s': 'alice', '-timeout': '20s', '-r': '1', '-m': '6', '-l': '6'} }
+properties:
+ dependencies:
+ - sipp :
+ version : 'v3.0'
+ - asterisk : 'chan_pjsip'
+
+ tags:
+ - pjsip
+ - RTP
\ No newline at end of file
diff --git a/tests/rtp/rtp_range/rtp_range_odd_odd/configs/ast1/extensions.conf b/tests/rtp/rtp_range/rtp_range_odd_odd/configs/ast1/extensions.conf
new file mode 100644
index 0000000..87291fb
--- /dev/null
+++ b/tests/rtp/rtp_range/rtp_range_odd_odd/configs/ast1/extensions.conf
@@ -0,0 +1,6 @@
+[default]
+
+exten => test,1,Answer
+exten => test,n,Noop(RTCP Range Test ASTERISK-27406)
+exten => test,n,Wait(15)
+exten => test,n,Hangup
diff --git a/tests/rtp/rtp_range/rtp_range_odd_odd/configs/ast1/pjsip.conf b/tests/rtp/rtp_range/rtp_range_odd_odd/configs/ast1/pjsip.conf
new file mode 100644
index 0000000..bef6c2f
--- /dev/null
+++ b/tests/rtp/rtp_range/rtp_range_odd_odd/configs/ast1/pjsip.conf
@@ -0,0 +1,19 @@
+[global]
+type=global
+debug=yes
+
+[local]
+type=transport
+bind=0.0.0.0
+protocol=udp
+
+[alice]
+type=endpoint
+context=default
+allow=!all,ulaw
+transport=local
+direct_media=no
+
+[alice]
+type=aor
+contact=sip:alice at 127.0.0.1:5061
diff --git a/tests/rtp/rtp_range/rtp_range_odd_odd/configs/ast1/rtp.conf b/tests/rtp/rtp_range/rtp_range_odd_odd/configs/ast1/rtp.conf
new file mode 100644
index 0000000..f791bc0
--- /dev/null
+++ b/tests/rtp/rtp_range/rtp_range_odd_odd/configs/ast1/rtp.conf
@@ -0,0 +1,3 @@
+[general]
+rtpstart=9999
+rtpend=10005
diff --git a/tests/rtp/rtp_range/rtp_range_odd_odd/sipp/call_accept_test_range.xml b/tests/rtp/rtp_range/rtp_range_odd_odd/sipp/call_accept_test_range.xml
new file mode 100644
index 0000000..68276a5
--- /dev/null
+++ b/tests/rtp/rtp_range/rtp_range_odd_odd/sipp/call_accept_test_range.xml
@@ -0,0 +1,117 @@
+<?xml version="1.0" encoding="ISO-8859-1" ?>
+<!DOCTYPE scenario SYSTEM "sipp.dtd">
+
+<scenario name="INVITE to echo with SDP in initial INVITE">
+ <send retrans="500">
+ <![CDATA[
+
+ INVITE sip:test@[remote_ip]:[remote_port] SIP/2.0
+ Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
+ From: test[call_number] <sip:[service]@[local_ip]:[local_port]>;tag=[call_number]
+ To: test <sip:test@[remote_ip]:[remote_port]>
+ Call-ID: [call_id]
+ CSeq: 1 INVITE
+ Contact: <sip:test@[local_ip]:[local_port];transport=[transport]>
+ Max-Forwards: 70
+ Subject: Test
+ User-Agent: Test
+ Content-Type: application/sdp
+ Content-Length: [len]
+
+ v=0
+ o=phoneA 53655765 2353687637 IN IP[local_ip_type] [local_ip]
+ s=-
+ c=IN IP[media_ip_type] [media_ip]
+ t=0 0
+ m=audio 6000 RTP/AVP 0
+ a=rtpmap:0 PCMU/8000
+
+ ]]>
+ </send>
+
+ <recv response="100"
+ optional="true">
+ </recv>
+
+ <recv response="488"
+ optional="true"
+ next="1"
+ rtd="true">
+ <action>
+ <ereg regexp="test[4-6]" search_in="hdr" header="From: " assign_to="4"/>
+ <test assign_to="4" variable="4" compare="equal" value=""/>
+ </action>
+ </recv>
+
+ <recv response="200" rtd="true">
+ <action>
+ <ereg regexp="m=audio 1000[0-4] RTP/AVP 0+..*"
+ search_in="body" check_it="true" assign_to="5"/>
+ <test assign_to="5" variable="5" compare="equal" value=""/>
+ </action>
+ </recv>
+
+ <send>
+ <![CDATA[
+
+ ACK sip:test@[remote_ip]:[remote_port] SIP/2.0
+ Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
+ From: test[call_number] <sip:[service]@[local_ip]:[local_port]>;tag=[call_number]
+ To: test <sip:test@[remote_ip]:[remote_port]>[peer_tag_param]
+ Call-ID: [call_id]
+ CSeq: 1 ACK
+ Contact: <sip:test@[local_ip]:[local_port];transport=[transport]>
+ Max-Forwards: 70
+ Subject: Test
+ Content-Length: 0
+
+ ]]>
+ </send>
+
+ <recv request="BYE">
+ </recv>
+
+ <send next="2">
+ <![CDATA[
+
+ SIP/2.0 200 OK
+ [last_Via:]
+ [last_From:]
+ [last_To:]
+ [last_Call-ID:]
+ [last_CSeq:]
+ Contact: <sip:[local_ip]:[local_port];transport=[transport]>
+ Content-Length: 0
+
+ ]]>
+ </send>
+
+ <label id="1"/>
+
+ <send next="2">
+ <![CDATA[
+
+ ACK sip:test@[remote_ip]:[remote_port] SIP/2.0
+ Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
+ From: test[call_number] <sip:[service]@[local_ip]:[local_port]>;tag=[call_number]
+ To: test <sip:test@[remote_ip]:[remote_port]>[peer_tag_param]
+ Call-ID: [call_id]
+ CSeq: 1 ACK
+ Contact: <sip:test@[local_ip]:[local_port];transport=[transport]>
+ Max-Forwards: 70
+ Subject: Test
+ Content-Length: 0
+
+ ]]>
+ </send>
+
+ <label id="2"/>
+
+ <!-- definition of the response time repartition table (unit is ms) -->
+ <ResponseTimeRepartition value="10, 20, 30, 40, 50, 100, 150, 200"/>
+
+ <!-- definition of the call length repartition table (unit is ms) -->
+ <CallLengthRepartition value="10, 50, 100, 500, 1000, 5000, 10000"/>
+
+</scenario>
+
diff --git a/tests/rtp/rtp_range/rtp_range_odd_odd/test-config.yaml b/tests/rtp/rtp_range/rtp_range_odd_odd/test-config.yaml
new file mode 100644
index 0000000..8256506
--- /dev/null
+++ b/tests/rtp/rtp_range/rtp_range_odd_odd/test-config.yaml
@@ -0,0 +1,29 @@
+testinfo:
+ summary: 'Verify Asterisk does not go inton an infinite loop and uses correct range'
+ description: |
+ This test verifies that the correct RTP Range is used and that Asterisk does not go
+ in to an infinite loop searching for a free RTP port
+ issues:
+ - jira: 'ASTERISK-27406'
+
+test-modules:
+ add-test-to-search-path: True
+ test-object:
+ config-section: 'test-object-config'
+ typename: 'sipp.SIPpTestCase'
+
+
+test-object-config:
+ memcheck-delay-stop: 7
+ test-iterations:
+ - scenarios:
+ - { 'key-args': {'scenario': 'call_accept_test_range.xml', '-i': '127.0.0.1', '-p': '5061', '-s': 'alice', '-timeout': '20s', '-r': '1', '-m': '6', '-l': '6'} }
+properties:
+ dependencies:
+ - sipp :
+ version : 'v3.0'
+ - asterisk : 'chan_pjsip'
+
+ tags:
+ - pjsip
+ - RTP
\ No newline at end of file
diff --git a/tests/rtp/rtp_range/tests.yaml b/tests/rtp/rtp_range/tests.yaml
new file mode 100644
index 0000000..f39df01
--- /dev/null
+++ b/tests/rtp/rtp_range/tests.yaml
@@ -0,0 +1,6 @@
+# Enter tests here in the order they should be considered for execution:
+tests:
+ - test: 'rtp_range_even_even'
+ - test: 'rtp_range_even_odd'
+ - test: 'rtp_range_odd_even'
+ - test: 'rtp_range_odd_odd'
diff --git a/tests/rtp/tests.yaml b/tests/rtp/tests.yaml
index 56e9f9e..2e2a177 100644
--- a/tests/rtp/tests.yaml
+++ b/tests/rtp/tests.yaml
@@ -2,3 +2,4 @@
tests:
- dir: 'strict_rtp'
- test: 'ast-2017-012'
+ - dir: 'rtp_range'
--
To view, visit https://gerrit.asterisk.org/c/testsuite/+/16636
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Gerrit-Project: testsuite
Gerrit-Branch: 16
Gerrit-Change-Id: Iefa8545cc1785b1e32a42d740f8f567706c73199
Gerrit-Change-Number: 16636
Gerrit-PatchSet: 1
Gerrit-Owner: Michael Bradeen <mbradeen at sangoma.com>
Gerrit-MessageType: newchange
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